The present invention relates to a technology for compression/expansion methods that can be applied to all transform based audio coders, or any audio coder using a windowing stage.
In 1991, the Motion Pictures Experts Group (MPEG), a group developed under the International Standards Organization(ISO), created an audio video system standard MPEG-1. MPEG-1 had three xe2x80x98layersxe2x80x99, the first two layers 1 and 2 were more simple audio coding and decoding algorithms, whereas the third layer, named MP3, was a much more complex audio coding and decoding system which just recently, has received alot of notoriety. MPEG-1 is a mono channel, stereo standard which operates at a 32-48 khz sampling rate. Around 1994, MPEG-2 was created which comprised of the same three layers, but this time was multichannel, or otherwise named (5.1) for 5 directions and one sub-woofer: Center, Left, Right, Left Surroundsound, Right Surroundsound and Low Frequency Exciter(LFE). MPEG-2 also operated at a much lower sampling rate, 12-32 kHz versus the 32-48 kHz of MPEG-1. In addition, MPEG-2 was backward compatible (BAC) with MPEG-1, which meant that MPEG-2 could play all MPEG-1 data streams.
More recently, around 1997, the thinking was that the audio coding and decoding standard could be made much more optimum if the standard did not have to be backward compatible (BAC). As a result, the audio coding and decoding standard, MPEG-2 non-backward compatible (NBC) was developed and it, as the name implies, was not backward compatible with the previous standards, MPEG-1 and MPEG-2. This standard was not commercially desirable(because of the xe2x80x98non-backward compatiblexe2x80x99 in the name) and so was changed to MPEG-2 Advanced Audio Coding (MC). MPEG-2 AAC is a multichannel system of up to 48 channels(foreign language applications are now enabled) and has a mono equivalence, if comparing against a mono standard like MP3(the third layer of MPEG-1) of 64 kbps versus MP3@64 kbps.
In any transform based audio decoder, there is a final xe2x80x9cwindow-overlap-addxe2x80x9d stage that converts the decompressed data into time domain output samples. The main data requirements to implement this stage are an input buffer containing the current decompressed data, a state buffer containing the previous decompressed data, and a constant table storing the xe2x80x9cwindowxe2x80x9d coefficients. These window-tables directly effect the quality of the output signal, and in order to keep this quality high, the tables require a significant amount of storage, about 2-4 k. In addition many of the audio compression algorithms provide support for multiple window shapes, so the storage requirements can increase to 4-8 k or more. In embedded applications, where memory is very limited, reducing the size of these tables is a necessity.
According to the present invention, there is developed a proprietary technology for compressing the window tables of audio coders to xe2x85x9 their original size (or less) without any loss of quality. This technology can be applied to all transform based audio coders, or any audio coder that uses a windowing stage. The novel technique for reducing storage requirements for the window tables of audio coders is based on multiple differentiation. Since the difference between any two adjacent window samples is relatively small, it is more efficient to store this difference. This technique can be carried out several more times, until the returns get smaller, and the computational requirements to xe2x80x9cundoxe2x80x9d the compression go up. The optimum number of times to differentiate is dependent on the particular application and the window shape.